blob: c480e37239519bf52afadf8df2d28053eed63206 [file] [log] [blame]
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
#include <memory>
#include <queue>
#include <set>
#include <vector>
#include "absl/base/attributes.h"
#include "api/rtp_packet_info.h"
#include "api/video/encoded_image.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/numerics/sequence_number_util.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace video_coding {
class PacketBuffer {
struct Packet {
Packet() = default;
Packet(const RtpPacketReceived& rtp_packet,
const RTPVideoHeader& video_header,
int64_t ntp_time_ms,
int64_t receive_time_ms);
Packet(const Packet&) = delete;
Packet(Packet&&) = delete;
Packet& operator=(const Packet&) = delete;
Packet& operator=(Packet&&) = delete;
~Packet() = default;
VideoCodecType codec() const { return video_header.codec; }
int width() const { return video_header.width; }
int height() const { return video_header.height; }
bool is_first_packet_in_frame() const {
return video_header.is_first_packet_in_frame;
bool is_last_packet_in_frame() const {
return video_header.is_last_packet_in_frame;
// If all its previous packets have been inserted into the packet buffer.
// Set and used internally by the PacketBuffer.
bool continuous = false;
bool marker_bit = false;
uint8_t payload_type = 0;
uint16_t seq_num = 0;
uint32_t timestamp = 0;
// NTP time of the capture time in local timebase in milliseconds.
int64_t ntp_time_ms = -1;
int times_nacked = -1;
rtc::CopyOnWriteBuffer video_payload;
RTPVideoHeader video_header;
RtpPacketInfo packet_info;
struct InsertResult {
std::vector<std::unique_ptr<Packet>> packets;
// Indicates if the packet buffer was cleared, which means that a key
// frame request should be sent.
bool buffer_cleared = false;
// Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
PacketBuffer(Clock* clock, size_t start_buffer_size, size_t max_buffer_size);
InsertResult InsertPacket(std::unique_ptr<Packet> packet)
InsertResult InsertPadding(uint16_t seq_num) ABSL_MUST_USE_RESULT;
void ClearTo(uint16_t seq_num);
void Clear();
// Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
absl::optional<int64_t> LastReceivedPacketMs() const;
absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
Clock* const clock_;
// Tries to expand the buffer.
bool ExpandBufferSize() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
// Test if all previous packets has arrived for the given sequence number.
bool PotentialNewFrame(uint16_t seq_num) const
// Test if all packets of a frame has arrived, and if so, returns packets to
// create frames.
std::vector<std::unique_ptr<Packet>> FindFrames(uint16_t seq_num)
void UpdateMissingPackets(uint16_t seq_num)
rtc::CriticalSection crit_;
// buffer_.size() and max_size_ must always be a power of two.
const size_t max_size_;
// The fist sequence number currently in the buffer.
uint16_t first_seq_num_ RTC_GUARDED_BY(crit_);
// If the packet buffer has received its first packet.
bool first_packet_received_ RTC_GUARDED_BY(crit_);
// If the buffer is cleared to |first_seq_num_|.
bool is_cleared_to_first_seq_num_ RTC_GUARDED_BY(crit_);
// Buffer that holds the the inserted packets and information needed to
// determine continuity between them.
std::vector<std::unique_ptr<Packet>> buffer_ RTC_GUARDED_BY(crit_);
// Timestamp of the last received packet/keyframe packet.
absl::optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
absl::optional<int64_t> last_received_keyframe_packet_ms_
absl::optional<uint32_t> last_received_keyframe_rtp_timestamp_
absl::optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
// Indicates if we should require SPS, PPS, and IDR for a particular
// RTP timestamp to treat the corresponding frame as a keyframe.
const bool sps_pps_idr_is_h264_keyframe_;
} // namespace video_coding
} // namespace webrtc